Issues 02000 - 02999

[..]
ASTERISK-02000: [post-1.0][patch] Custom command line for musiconhold software execution
ASTERISK-02001: AGI script does not see first digit from SIP cahnnel (using DTMF via SIP INFO)
ASTERISK-02002: Zaptel linux26 compile looks for /usr/src/linux-2.6/arch/x86 instead of /arch/i386
ASTERISK-02003: mobile: number not assigned/tmp. unavail: gives no event
ASTERISK-02004: Problem with Sip UNKN (d)
ASTERISK-02005: [patch] Notify on EOL of sip.conf incominglimit/outgoinglimit
ASTERISK-02006: [patch] Enhancement to "ParkedCalls" action in astman
ASTERISK-02007: cannot transfer calls placed on PSTN trunk
ASTERISK-02008: [patches][src-audit] apps directory files app_a*.c through app_m*.c
ASTERISK-02009: [PATCH] Teach RPM not to replace config files that already exist
ASTERISK-02010: [CONTRIB] Prepaid billing and rating apps
ASTERISK-02011: [patches][src-audit] apps directory files app_q*.c through app_z*.c
ASTERISK-02012: [patches][src-audit] directories: astman, cdr, db1-ast, editline, pbx, res, and stdtime
ASTERISK-02013: [patch] res_parking.c missing an include--get compiler warning
ASTERISK-02014: Strange behavior of zaptel T100p
ASTERISK-02015: RFC2833 + app_senddtmf
ASTERISK-02016: [patch] Different dialplan for caller number
ASTERISK-02017: RFC2833 + app_senddtmf
ASTERISK-02018: RFC2833 + app_senddtmf
ASTERISK-02019: ASTCC permission problem
ASTERISK-02020: Problems with insecure= setting and authentication
ASTERISK-02021: [patch][src-audit] channels directory -- last one
ASTERISK-02022: Coredump when waiting stream
ASTERISK-02023: [post-1.0][patch] Added Support to map table and cols in conf files for IAX2, SIP and VM
ASTERISK-02024: [patch] pbx.c: Code formatting and help text updates
ASTERISK-02025: [patch] assign more group for any channel (app_groupcount)
ASTERISK-02026: [patch] SIP registers outbound at wrong port
ASTERISK-02027: [post-1.0][patch] Sending HTML voicemail messages
ASTERISK-02028: [patch] app_voicemail is unable to update config in res_config_odbc
ASTERISK-02029: [PATCH] Tell RPM to put header files in a separate -devel.rpm
ASTERISK-02030: [patch] Update of ast_expr to use flex scanner - spaces no longer necc.
ASTERISK-02031: asterisk.c - not releasing allocated CLI memory
ASTERISK-02032: SIP SUBSCRIBE - NOTIFY Wrong Call-ID Match
ASTERISK-02033: mgcp reload stops asterisk from monitoring mgcp socket
ASTERISK-02034: Really long first ring, then normal
ASTERISK-02035: [patch] Let chan_zap compile when there is no libpri
ASTERISK-02036: [patch] Makefile error for chan_h323 linking
ASTERISK-02037: [patch] 7960 rejecting from field with :0 port description
ASTERISK-02038: [patch] Queue: new event when ringing phone, before bridging
ASTERISK-02039: [FreeBSD Only] Coredump in var resolving/substituting
ASTERISK-02040: bad checksum udp packets on linux
ASTERISK-02041: [patch] Some enhancements for astcc
ASTERISK-02042: externnotify script is always called even with no voicemail messages
ASTERISK-02043: [Request] Voicemail subscriber cannot send a new message
ASTERISK-02044: generating unique string for sending emails
ASTERISK-02045: /proc/zaptel/xxx doesn't work properly under Linux 2.6
ASTERISK-02046: pri channels lock after a few days, asterisk restart doesn't clear, needs zaptel restart
ASTERISK-02047: [patch] pbx.c: add skip and noanswer option to Background command
ASTERISK-02048: ast_dtmf_stream fails to return correct values
ASTERISK-02049: [post-1.0] [patch] An application that will help a user setup their voicemail account
ASTERISK-02050: [request] Privacy flag in SetCallerID, SetCIDNum, SetCIDName
ASTERISK-02051: [patch] RedHat asterisk init script replacement
ASTERISK-02052: [patch] res/Makefile contains line referencing parking.h which has been renamed
ASTERISK-02053: Missing DNS SRV support for SIP registration
ASTERISK-02054: SIP generated inband DTMF too short
ASTERISK-02055: SIP Re-invite still broken on some phones
ASTERISK-02056: callerid restriction list
ASTERISK-02057: [patch] Add command line arguments to safe_asterisk
ASTERISK-02058: [patch] chan_zap.c compiler warning / possible wrong value to libpri function
ASTERISK-02059: dsp.c fails to compile
ASTERISK-02060: dsp.c fails to compile
ASTERISK-02061: RTP.c not compiling on FreeBSD
ASTERISK-02062: MGCP wildcard endpoint audit fails
ASTERISK-02063: [Design & patch] app_privacy & confused users
ASTERISK-02064: [patch] Two minor Makefile-related patched and a useful addition to frame.c
ASTERISK-02065: Over amplified echo at start of call
ASTERISK-02066: [patch] Allow rejected VoIP dial to return to dialplan
ASTERISK-02067: [patch] Add return value check of ast_smoother_feed() before ast_smoother_read()
ASTERISK-02068: [patch] Updated help text for dial()
ASTERISK-02069: Improved counters to aid T1/E1 troubleshooting
ASTERISK-02070: Tiny patch to make app_dial return "CANCEL" when user press *
ASTERISK-02071: Changing useragent requires a restart
ASTERISK-02072: Registering with an intertex ix66 no longer works
ASTERISK-02073: Kernel Panic in Linux 2.6 when connection is broken to asterisk server
ASTERISK-02074: [patch] Missing return in chan_h323, function connection_made
ASTERISK-02075: Patch to clean many warnings in compilation of chan_h323
ASTERISK-02076: [patch] 'Newexten' manager event additional information
ASTERISK-02077: Zaptel or Libpri: PRI protocol error reseting calls
ASTERISK-02078: iaxy with one-way audio
ASTERISK-02079: Avoided deadlock
ASTERISK-02080: E1 stopped working in recent CVS.
ASTERISK-02081: Installing Asterisk from CVS on YDL 3.0.1 error
ASTERISK-02082: [FreeBSD 5.2.1. Only] Music On Hold fork blocks other threads
ASTERISK-02083: iaxy loses registrations
ASTERISK-02084: [*BSD only?] Invalid poll() processing, channel.c; Packets with bad UDP checksum.
ASTERISK-02085: [patch] simple ast_log()-like debugging
ASTERISK-02086: In build after 7/06/04 MWI on ADSI phones do not light up. But get shuttle tone
ASTERISK-02087: [post-1.0] [patch] Allow Caller TON to be retrieved in the dialplan
ASTERISK-02088: [patch] Remove quotation marks around MD5 arg in digest auth
ASTERISK-02089: [Patch] add manager events in sip and IAX. Back port from chan_sip2x.c.
ASTERISK-02090: [patch] [post 1.0] Improved SIP friends, supported postgres
ASTERISK-02091: [patch] Make astman compile on FreeBSD 4.9
ASTERISK-02092: [patch] memset on possible NULL pointer
ASTERISK-02093: [request] carrier grade CDR features requested
ASTERISK-02094: Dlink doesn't provide tone when handset is picked up
ASTERISK-02095: [Post-1.0][patch] Allow adding/removing queue members via manager API
ASTERISK-02096: X100P driver crashes 4kstacks kernel
ASTERISK-02097: [patch] add/remove members to/from queues on cli
ASTERISK-02098: [patch] new cdr module using asterisk manager
ASTERISK-02099: G.726 endianness not RFC compliant
ASTERISK-02100: DTMF in voicemail not completely recognized
ASTERISK-02101: [patch] Adding Belgium tones data
ASTERISK-02102: [patch] debug peer port correction
ASTERISK-02103: [patch] Port to Darwin 6.8/ OS X 10.3
ASTERISK-02104: Caller ID on FXS channels with Distinctive Ring
ASTERISK-02105: SIP Cancels Fail due to Request-URI Mismatch
ASTERISK-02106: [patch] ADSIProg() fails on exactly 245 bytes
ASTERISK-02107: outgoing sip calls ACK timeout error (wrong call-id) NAT server
ASTERISK-02108: chan_h323 & h.323 trace 2 and higher
ASTERISK-02109: ast_verbose() duplicate log messages if there is no "\n" at the end of the string.
ASTERISK-02110: DTMF stops working in Voicemail
ASTERISK-02111: [post-1.0] [request] MGCP Support As MG/SG, not CA/MG/SG
ASTERISK-02112: [patch] fxshonormode fix
ASTERISK-02113: [request] Pager e-mail from field
ASTERISK-02114: When using Meetme(<conf #>), won't retry if you enter incorrect pin
ASTERISK-02115: [new app] [post-1.0] app_sql_mysql
ASTERISK-02116: [patch] chan_alsa doesn't transmit, ignores configuration
ASTERISK-02117: [patch] Fail to get redirecting number
ASTERISK-02118: [patch] fix date values
ASTERISK-02119: Dial always goes to busy, never to unavailable
ASTERISK-02120: Add DNID to CDR
ASTERISK-02121: [request] Add ast_dsp to generate manager events for talk/silence.
ASTERISK-02122: [request] Be able to play audio files to an entire conference or specific user.
ASTERISK-02123: [request] Exit Meetme on ANY DTMF key
ASTERISK-02124: [request] Dispatch to extension based on *[0-9]
ASTERISK-02125: [request] Play audio file in app_meetme if user gets voicemail
ASTERISK-02126: [request] Specify ZapBarge audio direction
ASTERISK-02127: [request] ZapBarge - Ability to change channel without exiting zapbarge
ASTERISK-02128: [patch] remove option "outgoinglimit" that doesn't work anyway
ASTERISK-02129: issues with wrapuptime
ASTERISK-02130: RTP stream sent from wrong IP address if SIP address set to secondary IP address.
ASTERISK-02131: IAX phones cannot transfer some outgoing channels
ASTERISK-02132: [patch] Manager command ZapDialOffhook crashes when channel doesn't exist
ASTERISK-02133: Remove the cdr_pgsql.conf + Asterisk reload = Crash (Segmentation fault)
ASTERISK-02134: [patch] Add username to "sip show peer"
ASTERISK-02135: Slash operator to match extension based on callerid does not play well with changing callerid
ASTERISK-02136: fax detect does not detect fax from Brother Intellafax 4100
ASTERISK-02137: Modified some calls to permit functionality with Microsoft SQL Server
ASTERISK-02138: [patch] MOH pthread rather than fork
ASTERISK-02139: [request + patch] have call queues using a SIP member honor a 302 redirect
ASTERISK-02140: Cisco 7960 SIP incoming call issue
ASTERISK-02141: [patch] Show manager command required privileges in output of 'show manager commands'
ASTERISK-02142: [patch] Adding and removing agents with manager command should require Agent privs
ASTERISK-02143: Zap channel permanantly in conference after transferring to meetme
ASTERISK-02144: [patch] Allow priority to be set even if non-root user ID is specified
ASTERISK-02145: [patch] Allow the command-line editor to be chosen
ASTERISK-02146: [patch] Prevent blank lines from being saved in the history
ASTERISK-02147: Changes in Translation Path
ASTERISK-02148: [patch] Exit cleanly from remote control mode with SIGINT etc.
ASTERISK-02149: [patch] Default username from peer entry when creating new IAX2 channel
ASTERISK-02150: [PATCH] H.323 Memory corruptions
ASTERISK-02151: Limit number of calls to Agent
ASTERISK-02152: chan_mgcp.c does not support PING event
ASTERISK-02153: correct way to set wildcard endpoint in mgcp.conf
ASTERISK-02154: [request] res_musiconhold be reload enabled
ASTERISK-02155: [patch] allows specification of digittimeout when dialing extension during tranfers
ASTERISK-02156: patch from 2174 wrong fix
ASTERISK-02157: [patch] retrieve zapata channel status through asterisk manager
ASTERISK-02158: Uncompressed version of asterisk/sounds and asterisk-sounds/sounds
ASTERISK-02159: "mgcp reload" VERY buggy when mgcp.conf has wcardep= defigned
ASTERISK-02160: LOCAL_USER_ADD
ASTERISK-02161: translate.h using c++
ASTERISK-02162: ast_rtp_senddigit is hardcoded to use payload type 101
ASTERISK-02163: DTMF to SIP channel sent too soon
ASTERISK-02164: [patch] Fix compile failure on OpenBSD 3.5 (Release)
ASTERISK-02165: cdr_tds won't compile with older versions of FreeTDS
ASTERISK-02166: [report + patch] Forwarding issue with multiple dial targets
ASTERISK-02167: cdr_sqlite.so fail to create cdr.db+table
ASTERISK-02168: Asterisk does not hang up SIP call
ASTERISK-02169: inet_addr is not good to find out gw name is dotted ip or not ...
ASTERISK-02170: 'mgcp reload' always reloads wildcard endpoints reguardless if they were loaded on startup
ASTERISK-02171: When making a dynamic conference, no protection against duplicate conference numbers
ASTERISK-02172: failed to reset conferencing
ASTERISK-02173: Fix for bug #2200 doesn't check for dynamic_pin being null pointer
ASTERISK-02174: [patch] Add user number to MeetmeEnter and MeetmeLeave events
ASTERISK-02175: [post-1.0] [patch] h323-to-h323 and h323-to-mgcp native bridging
ASTERISK-02176: [docs patch] cdr documentation, well a start
ASTERISK-02177: No Ringback for ingress PSTN calls
ASTERISK-02178: [patch] add call timer on cli show channel and manager status
ASTERISK-02179: [patch] introduce the 'f' flag in app_directory (Directory()) to use the first name as a match
ASTERISK-02180: [patch] Make the ADSI feature download number and security code configurable
ASTERISK-02181: [patch] bugfix: asterisk may send SIP UA codec not offered in INVITE
ASTERISK-02182: [patch] Announce parking extension on ADSI compatable CPE
ASTERISK-02183: Return expression for wav_tell() wrong in format_wav_gsm.c
ASTERISK-02184: [patch] Verbose app
ASTERISK-02185: SIGBUS on sparc64/Linux
ASTERISK-02186: No ringing when I dial
ASTERISK-02187: [patch] * does not send back 'request identifier' in 'request notify' when wildcard endpoint sends 'notify'
ASTERISK-02188: [patch] Fix verbose output to not ignore replace parameter
ASTERISK-02189: wcfxs not detecting incoming calls
ASTERISK-02190: bug in dsp.c with silence detection
ASTERISK-02191: Avoid deadlock (2)
ASTERISK-02192: [patch] Avoid duplicate IP addresses/registrations in sip_friends database
ASTERISK-02193: call limit L(...) reset to initial value everytime something is dialed.
ASTERISK-02194: [patch] From, To, etc. messages do not handle < in quoted-strings
ASTERISK-02195: [patch] Asterisk sends corrupt data when peer dynamically switches from GSM to ULAW
ASTERISK-02196: Dead Lock on IAX2 start from 01 August to today using IaxComm to call
ASTERISK-02197: If videosupport=yes, SDP response on "voice only" calls include TWO media streams in SDP (one for voice and one for video)
ASTERISK-02198: Internal clocking on a 2nd of 2 TE410P boards
ASTERISK-02199: problem with Feature Group D / E&M Wink
ASTERISK-02200: Tab Completion Repeating
ASTERISK-02201: [Patch] correct MeetMe for Marked users
ASTERISK-02202: Please provide PRI_HANGUPCAUSE
ASTERISK-02203: Crash on CVS
ASTERISK-02204: [patch] Allow use of on/off in ast_true() for validation of config options
ASTERISK-02205: [patch] create dynamically conference with the CLI
ASTERISK-02206: negative lag causes jitter buffer to grow larger than maxexcessjitter setting
ASTERISK-02207: [request] make generators work even if no received audio available
ASTERISK-02208: SNDCTL_DSP_SETDUPLEX does not work
ASTERISK-02209: Extension 'T' (AbsoluteTiemout) inside a macro does not work as expected.
ASTERISK-02210: ADSI voicemail folders softkeys display incorrectly
ASTERISK-02211: [patch] [post-1.0] allow TON, NPI and Presentation to be retrieved in the dialplan
ASTERISK-02212: [patch] MGCP does not currently have support for pre-rfc mode "draft" operation or future rfc definitions
ASTERISK-02213: [patch] manager getvar action missing double CRLF in Response
ASTERISK-02214: [patch] Add ";user=phone" when INVITE contain only phone number
ASTERISK-02215: [report + patch] System() application troubles.
ASTERISK-02216: Please include indicator for in/outbound in CDR-table
ASTERISK-02217: MGCP call pickup with *8 and *8x options
ASTERISK-02218: Asterisk doubles DTMF events when endpoint in INBAND mode
ASTERISK-02219: [patch] [h.323] Auto dialing crashes
ASTERISK-02220: Channel variables in AGI application after Dial.
ASTERISK-02221: [patch] Implement fax detection for SIP calls.
ASTERISK-02222: chan_h323 memory leak ?
ASTERISK-02223: [patch] a swap is a swap, but
ASTERISK-02224: [patch] Using a stream for MusicOnHold
ASTERISK-02225: Missing statement of licence for included FPM mp3 files
ASTERISK-02226: [post 1.0] REFER transfer fails with certain hardware (REFER requires NOTIFY)
ASTERISK-02227: [patch] Problems with () characters in expressions in extensions.conf
ASTERISK-02228: A @ character as a username for a sip host is read wrong
ASTERISK-02229: [chan modem] Asterisk internal DTMF detection crashes
ASTERISK-02230: pbx*CLI> help dial is incomplete
ASTERISK-02231: [patch] iax2 ignores port and serverport values in iax.conf, iaxprov.conf
ASTERISK-02232: chan_zap allows you to bridge the two subchannels of a single master
ASTERISK-02233: sample in configs/extconfig.conf.sample is wrong
ASTERISK-02234: [patch] Empty messages should not result in a voicemail
ASTERISK-02235: Mp3Player (for asterisk) doesn't support http:// shoutcast streams
ASTERISK-02236: [patch] Reboot Grandstreams phones from CLI (proprietary NOTIFY)
ASTERISK-02237: [patch] Set/show MWI settings for Zap channels from CLI
ASTERISK-02238: SIGFPE causes asterisk to crash (i4l)
ASTERISK-02239: Does the new Bugs home work?
ASTERISK-02240: test 2
ASTERISK-02241: [request] make table name for "cdr" table configurable in cdr_odbc
ASTERISK-02242: res_config_odbc doesn't load full context if id aren't continous (extensions.conf)
ASTERISK-02243: agi stream file randomly exits with: "ast_waitstream_full: Wait failed (No such file or directory)" message.
ASTERISK-02244: Patch to properly build and run on NetBSD
ASTERISK-02245: [patch] h323-sip-mgcp native bridging
ASTERISK-02246: AGI command "stream file" produces Error on Fedora Core 2
ASTERISK-02247: [patch] Small memory leak on unregister of applications
ASTERISK-02248: [patch] Allow functions to be set
ASTERISK-02249: [patch] save dialplan does not store CID matching or switch commands
ASTERISK-02250: Latest cvs causes asterisk to crash
ASTERISK-02251: [patch] TDM400P FXO: Unexpected control subclass '5'
ASTERISK-02252: Intel modem 536EP problem
ASTERISK-02253: [probably not really a bug] IAX2 DTMF not recognized - CVS-HEAD-08/01/04-22:51:56
ASTERISK-02254: 1st second of RTP (especially indications) to sip channel is poor quality
ASTERISK-02255: Makefile has hardcoded paths, should always use paths in /etc/asterisk/asterisk.conf
ASTERISK-02256: rtptimeout and canreinvite=yes
ASTERISK-02257: [request] Port Restart
ASTERISK-02258: typo in astcc
ASTERISK-02259: extensions reload does not recognize timeout in extensions.conf by save dialplan
ASTERISK-02260: Turn Callwaiting off on IAXy boxes
ASTERISK-02261: [patch] show applications like <text>
ASTERISK-02262: Call parking does not return to original caller after timeout.
ASTERISK-02263: [patch] Unable to call X-lite using SpeeX
ASTERISK-02264: zap show channel : provide current call-duration
ASTERISK-02265: asterisk + mod_php like in Apache
ASTERISK-02266: [request] Enable option to use ICC compiler instread of GCC.
ASTERISK-02267: [patch] vmail.cgi modified to work with both voicemail.conf and MySQL voicemail configuration.
ASTERISK-02268: AGI get_variable doesn't work.
ASTERISK-02269: [patch] Destorying target span of DACS crashes the kernel
ASTERISK-02270: Digium FAQ page contains deprecated BYEXTENSION syntax
ASTERISK-02271: [patch] Add quick login to voicemailmain when user does not enter login
ASTERISK-02272: [patch] When playing a message longer than X minutes, say the duration in minutes
ASTERISK-02273: chan_sip does not pass digits while using nat
ASTERISK-02274: [Request] Please add zap timing method to rtp.c
ASTERISK-02275: [Patch] Add Confirm Answer (like in chan_zap) into app_dial
ASTERISK-02276: [patch] Rename events omit Uniqueid
ASTERISK-02277: app_sms: 1 char missing in directory name.
ASTERISK-02278: Uniden phones will not work behind a nat
ASTERISK-02279: callerid field in sip.conf changes Contact: header
ASTERISK-02280: [request] Allow Asterisk to send the Caller Name in the q931 Facility Message
ASTERISK-02281: I4L loops on a call that is hung up
ASTERISK-02282: channel.c 1.134 breaks MOH for MeetMe
ASTERISK-02283: [patch] chan_iax2.c Flexibility in MYSQL access
ASTERISK-02284: [patch] MeetMe option to place callers in MOH instead of Hangup when marked user (temporarily) leaves
ASTERISK-02285: nat=yes saves private ip after new nat=route patch
ASTERISK-02286: Problem in musiconhold.h while using C++
ASTERISK-02287: chan_sip.c: 7731 sip_poke_noanswer: Peer is now UNREACHABLE
ASTERISK-02288: Inbound SIP call Status
ASTERISK-02289: [request] Makefiles do not correctly recognize existence of UTRASPARC Family of CPU's.
ASTERISK-02290: Can only insert one CDR Record...
ASTERISK-02291: The bugtracker needs a 'forgotten password' feature
ASTERISK-02292: app_directory incorrectly refers to 'context' when 'dialcontext' has been defined
ASTERISK-02293: Division by zero error - SIGFPE crash
ASTERISK-02294: New Error Messages (Also Left Calls)
ASTERISK-02295: [Patch] Busy/Congestion apps don't work with PRI
ASTERISK-02296: Error when legacy PBX gets line from te405p
ASTERISK-02297: no EOF after AGI answer to command: "channel status"
ASTERISK-02298: Add a courtesy beep to indicate connection to party calling into a parked call
ASTERISK-02299: [PATCH]: Minor sparc optimiation in build system for libpri
ASTERISK-02300: [patch] Poke peer when we have IP
ASTERISK-02301: #define _THREAD_SAFE in localtime.c
ASTERISK-02302: [patch] Improved ultrasparc support for Asterisk
ASTERISK-02303: app_festival creates files with permissions set to 000
ASTERISK-02304: iaxys still losing registration
ASTERISK-02305: res_config_odbc, all selects fail
ASTERISK-02306: Exceptionally long queue length warnings being continuously repeated
ASTERISK-02307: [request] Put Span Out Of Service from CLI
ASTERISK-02308: [patch] Add verbosity to pbx.c when it can't find a target triplet in the dialplan
ASTERISK-02309: [patch] add verbosity to parking timeout event in res_features.c
ASTERISK-02310: [patch] postgres music on hold
ASTERISK-02311: Native bridging does Link/Unlink/Link
ASTERISK-02312: [patch] Bug in handle_add_queue_member
ASTERISK-02313: [request] immediate=yes / threeway call
ASTERISK-02314: [request] print to stderr in AGI should also show up in CLI when connecting via "asterisk -r"
ASTERISK-02315: [Patch] Add Support for the Local channel to astcc
ASTERISK-02316: chan_zap fails to build under uclibc
ASTERISK-02317: [patch] Makefile cleanup a bit...
ASTERISK-02318: Use count doesn't decrease in format_wav_gsm.c
ASTERISK-02319: [patch] add verbosity and a warning to app_voicemail
ASTERISK-02320: app_read.c outputs garbage in console
ASTERISK-02321: [patch] Add option to force immediate password change if user has a specific password
ASTERISK-02322: Members count of calls taken is wrong when *8 is used
ASTERISK-02323: [not an asterisk bug] Malformed 401 Message from SER with bindaddr=0.0.0.0 and asterisk coneccted to two local networks
ASTERISK-02324: [request] Add sound file playback to Dial Application option c
ASTERISK-02325: [patch] add externpass cmd and voicemail reload
ASTERISK-02326: [request] [post 1.2] support reponses to INVITEs on IP address aliases to come back on the same IP
ASTERISK-02327: TDM400P modules do not recognize lack of cable/dial tone
ASTERISK-02328: [patch] Add temporary greetings to voicemail
ASTERISK-02329: Please make TON (Type of Number) accessible in asterisk
ASTERISK-02330: Changes to chan_zap.c version 1.331 break channel bank FXO ports
ASTERISK-02331: [Patch] exitcontext option does not work if 'o' or 'a' not in current context
ASTERISK-02332: [patch] Make the codecs we answer with configurable
ASTERISK-02333: [patch] output time in hms a channel has been active when cli 'show channels' is called.
ASTERISK-02334: [patch] Add mysqlcanblock= and mysqlsipfriends= to chan_sip
ASTERISK-02335: missing symbol in latest cvs
ASTERISK-02336: Trouble with configuration files when entering ODBC connection strings
ASTERISK-02337: SIP 400 response "Missing/Invalid From" -> DIALSTATUS="NOANSER"
ASTERISK-02338: IAXY generates incorrect timestamps in IAX2 stream
ASTERISK-02339: no out bound audio not even on echo test
ASTERISK-02340: [patch] append hostname to logfiles
ASTERISK-02341: [patch] couple of voicemail password change bugs
ASTERISK-02342: g.729 show license is not working
ASTERISK-02343: [Feature + Patch] E-mail info and list for astcc
ASTERISK-02344: [Patch] Wait until # is press before making the call go throught
ASTERISK-02345: [patch] interrupt user and admin menu by choosing option
ASTERISK-02346: [Patch] Allow the use of L(x) flag in Background to play a specified language
ASTERISK-02347: [patch] Zap attached devices no long receive callerid
ASTERISK-02348: [patch] -t flag to asterisk args by anthm
ASTERISK-02349: [patch] add format_mp3, format_slinear and format_base64_wav_gsm by anthm
ASTERISK-02350: [Patch] Make w flag work better, and say prompt on enter/leave of marked
ASTERISK-02351: [patch] res_monitor patch by anthm
ASTERISK-02352: [post-1.0] [patch] res_sqlite, adds sqlite_switch, cdr engine, cli tools and SQL dialplan application by anthm
ASTERISK-02353: [patch] if no password set, don't bother asking
ASTERISK-02354: [patch] prevent a user changing mailbox password
ASTERISK-02355: [Patch] Switch from user to admin in meetme
ASTERISK-02356: alloca.h in utils
ASTERISK-02357: usedistinctiveringdetection in X100P analog cards
ASTERISK-02358: Compile failure in say.c since cvs version 1.35
ASTERISK-02359: SIP response 503 "Service Unavailable" -> DIALSTATUS = NOANSWER
ASTERISK-02360: Avoided deadlock for Zap channel
ASTERISK-02361: [general] General Holding bug for BSD/OSX Compatibility
ASTERISK-02362: Problem with DTMF being passed from Cisco GW to asterisk on ingress calls
ASTERISK-02363: [patch] callerid.c TELEKOM callerid
ASTERISK-02364: [patch] Modify level of log entry related to GotoIf when no branch taken
ASTERISK-02365: [patch] fix padding of field in iax2 debug output to aid readability
ASTERISK-02366: [request] [patch] Lightweight ODBC API for asterisk
ASTERISK-02367: When dialling #, asterisk gets %23
ASTERISK-02368: Call progress detection for Costa Rica is broken
ASTERISK-02369: rtptimeout hanging up "dialling channels"
ASTERISK-02370: One-way audio when transferring calls after they've been picked up by a queue
ASTERISK-02371: [PATCH TRIVIAL] Make action_getvar() respect ActionID.
ASTERISK-02372: [PATCH TRIVIAL] Make action_getvar() respect ActionID.
ASTERISK-02373: Asterisk Manager response corruption.
ASTERISK-02374: dynamic and default queue members call counters respond different to a reload.
ASTERISK-02375: Asterisk Deadlock
ASTERISK-02376: geting callingpres status
ASTERISK-02377: Macro support in Dial()
ASTERISK-02378: [patch] SIP Headers - Modified Format Handler
ASTERISK-02379: ODBC() application for asterisk.
ASTERISK-02380: Incorrect Destination written in CDR when Macro() used with Goto()command
ASTERISK-02381: ast_channel_walk_locked avoiding a deadlock causes breaks in audio
ASTERISK-02382: GSM sounds fail to play on GSM call
ASTERISK-02383: When leaving a VM, 'accepting' recording doesn't hang up on person.
ASTERISK-02384: app_mp3 used in MP3Player fails to play stream due to timeout
ASTERISK-02385: * dont know ipaddress of dynamic sip peers after restart
ASTERISK-02386: DIALSTATUS through IAX trunk?
ASTERISK-02387: Bug introduced in CVS with the guardtime feature
ASTERISK-02388: problem with sip registration (to sipgate.de) with pedantic=yes
ASTERISK-02389: [PATCH] Automate import of trunks
ASTERISK-02390: calls and audio recorded from GS phone plays back at double speed
ASTERISK-02391: Who would like agents to be able to respond to emails as well as calls?
ASTERISK-02392: [post-1.0][patch]Move 'lastcallerid' into 'struct ast_channel'
ASTERISK-02393: call-id header with FQDN or externip
ASTERISK-02394: [patch] Add 'set debug n' to cli and tweak 'set verbose n'
ASTERISK-02395: [post 1.0] [patch] ODBC voicemail, support of HTML messages
ASTERISK-02396: iax2 show channels report too great lag
ASTERISK-02397: NOTICEs from sched.c
ASTERISK-02398: ringing tones not transmitted
ASTERISK-02399: regexten feature improvement request
ASTERISK-02400: [patch] make manager originate send status in case of asynchronous execution.
ASTERISK-02401: Matching on dialstatus in the dialplan as laid out in extensions.conf.sample does not function.
ASTERISK-02402: voicemail file sequencing problem
ASTERISK-02403: Add possibility to enable/disable sip friends from sip.conf
ASTERISK-02404: Missing simicolon in latest chan_sip (CVS rev 1.502)
ASTERISK-02405: MYSQL Voice Mail Lookup
ASTERISK-02406: Karma changes (please ignore)
ASTERISK-02407: trunk not always used
ASTERISK-02408: [request] auto limit IAX2 trunked calls or autoincrease the value to avoid flooding warning messages.
ASTERISK-02409: after latest CVS update, MP3Player command giving RTP Bad Packet Errors
ASTERISK-02410: compatability problem with huawei sip interface.
ASTERISK-02411: [patch] channel.c 'timeleft' uses non-existant sound files
ASTERISK-02412: SIP and h323 can't handle LPC10
ASTERISK-02413: Call rejected -> DIALSTATUS=NOANSWER
ASTERISK-02414: RTP is not immediately transmitted after 183 session progress is sent
ASTERISK-02415: [post-1.0 patch] ANI II exposed as a VAR in dialplan.
ASTERISK-02416: app_disa no longer restores dialtone after ignorepat
ASTERISK-02417: Parking calls with blind transfer phones like GrandStream does not read digits
ASTERISK-02418: Including a 'w' in the dial string on a PRI trunk breaks calling number in Q.931 frame?
ASTERISK-02419: [patch] Pick-up extension is not configurable
ASTERISK-02420: [patch] strncpy with wrong sizeof in config parsing
ASTERISK-02421: "make update" don't use compression
ASTERISK-02422: Answer() can interfere with E&M Wink signalling on cT1
ASTERISK-02423: Asterisk does not stimulate a far end disconnect when Hangup() called on E&M trunk during handshake
ASTERISK-02424: [PATCH] Client application to check Usage Info
ASTERISK-02425: [patch] Attended Pound Transfers The Perfect New Years Resolution!!!
ASTERISK-02426: Authentication _sometimes_ fail with clients
ASTERISK-02427: Suggested clarification in app.h
ASTERISK-02428: [PATCH] Add advance call timeout warnings
ASTERISK-02429: [PATCH] Reset inuse info in card database
ASTERISK-02430: [patch] to call cli functions from agi script
ASTERISK-02431: [new app] WaitForSilence application
ASTERISK-02432: Segmentation fault in chan_iax2.c
ASTERISK-02433: MYSQL Voice Mail Lookup and IAX Friends
ASTERISK-02434: Unable to pass parameters of macro into Dial (from AGI script).
ASTERISK-02435: [patch] initial Remote-Party-ID support & app to set RestrictCID
ASTERISK-02436: [patch] Voicemail login causes change to invalid/corrupt language code
ASTERISK-02437: Manager Redirect action does not function in certain conditions.
ASTERISK-02438: [patch] Allow app_directory to work with REALTIME
ASTERISK-02439: [patch] vmail.cgi does not use MySQL database when enabled.
ASTERISK-02440: ZT_TIMERPING
ASTERISK-02441: Linking wrong threads library and other threads issues
ASTERISK-02442: With rpid "privacy=full" callerid is always "Unknown" <unknown>
ASTERISK-02443: [patch] term.c - support Eterm colors
ASTERISK-02444: No response -> NO ANSWER
ASTERISK-02445: the CALLERIDNAME information not populated on a PRI Call
ASTERISK-02446: asterisk SIP ignores SIP packets from peer in "early" state
ASTERISK-02447: Patch 0002434 can cause broken event output.
ASTERISK-02448: waw49 files sometimes get size for data chunk on record
ASTERISK-02449: App Sox (new app -anthm)
ASTERISK-02450: Data through TE410P stops for few seconds
ASTERISK-02451: [patch] Caller ID wrong on call transfer
ASTERISK-02452: res/res_agi.c doesn't compile
ASTERISK-02453: [patch] Eject last user
ASTERISK-02454: change in chan_mgcp.c causes dtmfmode=inband to not work
ASTERISK-02455: Update Mantis
ASTERISK-02456: [patch] Detect terminals which support color
ASTERISK-02457: rtp.c generates seemingly erroneous warning message
ASTERISK-02458: SIP NOTIFY returned during Supervised Transfer not according to RFC
ASTERISK-02459: [patch] AGI Debugging support
ASTERISK-02460: rtp.c warning - rtp structure is null
ASTERISK-02461: [patch] Holdtime incorrectly announced as '1 minutes' when holdtime less than 2 minutes
ASTERISK-02462: [patch] Fix 'show applications like <string>'
ASTERISK-02463: [discuss] Race condition in timed includes
ASTERISK-02464: [patch] Queue is not ejecting callers if all members are logged off.
ASTERISK-02465: An unavailable CODEC is sometimes accepted
ASTERISK-02466: Asterisk Manager Proxy -- simpleproxy.pl
ASTERISK-02467: [patch] "Thank you" message should not be played after first position announcement
ASTERISK-02468: app_saycount.pl.c - an application that returns the right counting word for a number in Polish
ASTERISK-02469: [patch] cdr.txt typo
ASTERISK-02470: [patch] Override position announcement when there is nobody waiting in the queue
ASTERISK-02471: Transferring Calls
ASTERISK-02472: [patch against 1.0] res_agi needs socket defs
ASTERISK-02473: compilation against latest ilbc don't work
ASTERISK-02474: [patch] cleaned up cdr_mysql.c
ASTERISK-02475: No voice on AT&T 4ess trunkgroup
ASTERISK-02476: Non d-channel T's not working on AT&T trunk
ASTERISK-02477: Provide configurable timers for PRI's
ASTERISK-02478: Ye old: 'Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)'
ASTERISK-02479: [patch] Send email messages from within asterisk
ASTERISK-02480: [request] SNOM 200 CMC code to UserData field in CDR
ASTERISK-02481: reload and extensions reload forget subscriptions for presence (e.g. SNOM busy lamps)
ASTERISK-02482: [request] Record calls at ZAP interface
ASTERISK-02483: Missing Curly braces appear to cause sip_reg_timeout problem (numerous SIP channels left open)
ASTERISK-02484: id3 version 2 tags on Free Play Music files crash asterisk 1.0 (and CVS) on gentoo
ASTERISK-02485: Asterisk does not respond to BYE message when context is set to canreinvite=Yes
ASTERISK-02486: [request] priority of same and next in extensions.conf
ASTERISK-02487: [request] Can MYSQL_LOGUNIQUEID be automatic...
ASTERISK-02488: Code cleanup, extensions including regex matching
ASTERISK-02489: [patch] Add channel group setting to Dial()
ASTERISK-02490: res_agi.c will not compile on FreeBSD
ASTERISK-02491: [patch] new jitter buffer
ASTERISK-02492: [patch] app_callback - calls users back.
ASTERISK-02493: app_math adds Sum, Multiply, Divide, Subtract, Modulus, GT, LT, GTE, LTE, EQ functions to asterisk
ASTERISK-02494: 2342 bug fix breaks chan_zap answer behavior
ASTERISK-02495: [patch] Adjustabe Speex Codec
ASTERISK-02496: Linux 2.6: Timing off when using USB devices
ASTERISK-02497: AGI network scripts (agi:// URLs) do not parse ports correctly
ASTERISK-02498: AGI network scripts (agi:// URLs) do not parse ports correctly
ASTERISK-02499: type-punned pointer will break strict-aliasing rules ... hash/ndbm.c out of date ?
ASTERISK-02500: [patch] remove warning from localtime.c
ASTERISK-02501: Design flaw in chan_sip
ASTERISK-02502: recent "less than" changes broken
ASTERISK-02503: [patch] Fix endless loop in card generation when starting digit=0
ASTERISK-02504: [Patch] Undocumented feature now documented
ASTERISK-02505: [patch] French update for say.c
ASTERISK-02506: [request] chan_zap.c:7358 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1
ASTERISK-02507: Ignore ;transport=udp from 302 Moved Temp. requests
ASTERISK-02508: [patch] IAX2 Native Bridge Crashes
ASTERISK-02509: No sound generated by asterisk (prompts)
ASTERISK-02510: Ignore message type 6 in CID delivery
ASTERISK-02511: SIP Early Media Not Working
ASTERISK-02512: [PATCH] Assorted additions - SIP&IAX Info
ASTERISK-02513: silence when logged in (second) via SIP from Grandstream HT 286 into Meetme
ASTERISK-02514: Transfer blocks new agent calls
ASTERISK-02515: app_vareval - allows the evaluation of dynamically built variables
ASTERISK-02516: STREAM FILE and GET DATA not work in current cvs
ASTERISK-02517: [External Module] res_sqlite for SQLite 3
ASTERISK-02518: Has(New)VoiceMail does not assume INBOX if no folder specified
ASTERISK-02519: NetBSD build doesn't always find libncurses
ASTERISK-02520: [patch] Rework ast_app_has_voicemail to handle more cases
ASTERISK-02521: extens with cidmatch no longer work on zap
ASTERISK-02522: [patch] Delay member connect to caller
ASTERISK-02523: [patch] Report caller's hold time to agent
ASTERISK-02524: [patch] MeetMe needs configurable Music On Hold
ASTERISK-02525: SIP_CODEC variable not working in 1.0.0 and 1.0.1
ASTERISK-02526: PRES_NUMBER_NOT_AVAILABLE is not defined in pri_pres2str
ASTERISK-02527: [patch] chan_sip.c will not properly compile with latest callerid changes and MYSQL support
ASTERISK-02528: [patch] ast_true in cdr_odbc
ASTERISK-02529: [patch]RTP debugging
ASTERISK-02530: chan_mgcp crashes * CVS-HEAD-09/29/04
ASTERISK-02531: Asterisk doesn't build with uclibc
ASTERISK-02532: app_queue incorrectly reports that no one is answering the queue
ASTERISK-02533: Asterisk will dergister all SIP phones but will respond to cli cmds
ASTERISK-02534: Latest CVS (2004-10-05 12:30) crashes if an attempt is made to transfer a call.
ASTERISK-02535: [patch] mysql log of queue_log
ASTERISK-02536: Allow "c" dial flag to work with non-zap channels
ASTERISK-02537: chan_mgcp binds rtp to 0.0.0.0 insted of addr, specified in "bindaddr" option
ASTERISK-02538: notification msg during a blind transfer does not include the registrar's IP address
ASTERISK-02539: Discontiguous descending Zap channel group fails to use all members
ASTERISK-02540: SIP MWI not working with Xten eyeBeam and possibly other clients
ASTERISK-02541: Audio from PBX not heard on Polycom IP500/SIP phone with current release or -HEAD
ASTERISK-02542: [poet 1.0] [new app] app_realtime
ASTERISK-02543: TRON TROFF in extensions.conf
ASTERISK-02544: [patch] Always fflush for any CDR file (not only for Master.csv)
ASTERISK-02545: garbled audio with trunking on some devices
ASTERISK-02546: Typos on Download page
ASTERISK-02547: RDNIS is always empty
ASTERISK-02548: chan_sip 1.521 breaks SIP NOTIFY
ASTERISK-02549: receiving gethostbyname() error on call to festival in dialplan
ASTERISK-02550: [patch] Count for show modules
ASTERISK-02551: [patch] voicemail beep shouldn't be played till we get next message number
ASTERISK-02552: [patch] memset fixes
ASTERISK-02553: Problem with MWI in the bugfix of chan_sip.v ver 1.520 - 1.521
ASTERISK-02554: [patch] app_sms doesn't use paths from asterisk.conf
ASTERISK-02555: attribution missing in CHANGES
ASTERISK-02556: [patch] Memory leak fixes for chan_sip.c with realtime() functions
ASTERISK-02557: ILBC produces choppy sound
ASTERISK-02558: [patch] Add mute & umute ALL
ASTERISK-02559: [patch] Fix two trivial verbosity related issues
ASTERISK-02560: [patch] Elaborate on where rejected IAX2 call was trying to reach
ASTERISK-02561: No Audio / Fast busy on incoming call on T100P
ASTERISK-02562: Nortel SIP dtmf
ASTERISK-02563: Developer documentation for realtime config
ASTERISK-02564: [devel branch] asterisk segfault on reload
ASTERISK-02565: cidmatch broken when pattern contains ranges
ASTERISK-02566: [patch] app_lookupcidname borken (CVS rev 1.4 of app_lookupcidname.c)
ASTERISK-02567: [patch] chan_iax2 shouldn't use IAX/Registry on temponly peers
ASTERISK-02568: Can't use virtual IP's for VOIP
ASTERISK-02569: [request] Detect dialtone on Zap channels before dialing
ASTERISK-02570: res_config_mysql - realtime driver
ASTERISK-02571: h323 channel make error
ASTERISK-02572: Patch to modify queue message system
ASTERISK-02573: chan_sip should reply FROM the same address that request was sent TO
ASTERISK-02574: [patch] set verbose and set debug still conflict with each other
ASTERISK-02575: [patch] don't seed p->temponly sipfriends when using realtime
ASTERISK-02576: [patch] Remove VoiceMail2 and VoiceMailMain2 backwards compat in cvs-head
ASTERISK-02577: SIP INVITE header doesn't include number to dial.
ASTERISK-02578: qualify != 'no' in realtime (sipfriends extconfig) make the peer unreachable and sip show peer PEER_NAME endup hanging Asterisk
ASTERISK-02579: All Dial commands fail after OSPLookup
ASTERISK-02580: [v1-0] Recent callToken changes break send_digit
ASTERISK-02581: Cisco BTS problem
ASTERISK-02582: [patch] app_realtime - add CLI debug abilities
ASTERISK-02583: [patch] terminator key for app_record
ASTERISK-02584: [patch] Add channel variable to BackgroundDetect / app_talkdetect
ASTERISK-02585: AGI Application tends to become non-responsive
ASTERISK-02586: chan_h323 doesn't initiate calls at all
ASTERISK-02587: [request] "no call progress" indication timeout option.
ASTERISK-02588: [patch] Fixes to support attended transfers from a libiax client
ASTERISK-02589: Record application fails with ast_set_read_format: Unable to find a path from GSM to UNKN
ASTERISK-02590: tab completion on show dialplan from -r asterisk with large extensions.conf will segfault
ASTERISK-02591: [patch] more details for sip show peers
ASTERISK-02592: [patch] preliminary zap reload from cli (no signalling, just parameters)
ASTERISK-02593: [patch] Possibility to disable some parts of the announce
ASTERISK-02594: [patch] Native MOH without mpg123
ASTERISK-02595: usage count isn't decremented when agent/queue call completed
ASTERISK-02596: SetLanguage without parameter segfault
ASTERISK-02597: Segfault when res_config_odbc and res_config_mysql are load
ASTERISK-02598: [patch] show module like <keyword>
ASTERISK-02599: [patch] Improved debug output in SIP and IAX (lagging, RSA key info)
ASTERISK-02600: [PATCH] Tweak 'pri debug' so output is displayed even when verbosity = 0
ASTERISK-02601: Directory Application doesn't work if using realtime config.
ASTERISK-02602: Voicemail configuration still pulls zone info from voicemail.conf if using realtime config.
ASTERISK-02603: Voicemail notification is not sent to e-mail/pager if using realtime config.
ASTERISK-02604: BSD systems fail with "bindaddr=0.0.0.0"
ASTERISK-02605: Dialing from keypad in gnomemeeting crashes asterisk
ASTERISK-02606: [request] presence/hint for queues
ASTERISK-02607: chan_h323 core dumps
ASTERISK-02608: GotoIfTime does not support new "n" priority
ASTERISK-02609: [patch] update to readme
ASTERISK-02610: [patch] Makefile tweak to point to kernel build in /lib/modules/VERSION/build
ASTERISK-02611: c option is not documented in show application dial
ASTERISK-02612: [patch] Update of README
ASTERISK-02613: Breakage of ChanIsAvail because of bad hangup handling...
ASTERISK-02614: Chan_H323 dials out incorrectly
ASTERISK-02615: Dial(Zap/1) without timeout gives dialtone
ASTERISK-02616: [patch] app_festival calls ast_destroy on cfg before it's done
ASTERISK-02617: sip stops communicating when gethostbyname() temporarily fails
ASTERISK-02618: tor2.c tor2_probe
ASTERISK-02619: [patch] fix oops on reload chan_zap.so if we're using pseudo channels.
ASTERISK-02620: [PATCH] Make operator=no feature of app_voicemail behave as documented
ASTERISK-02621: The version 1.29 of loader.c crash on the apllication app_qcall and makr undefined symbol
ASTERISK-02622: [isdn4linux] echotest / voicemail crash when receiving DTMF from the PSTN[
ASTERISK-02623: [patch] Use of bindaddr=0.0.0.0 broken under BSD
ASTERISK-02624: After #2650 fix chan_h323 doesn't build anymore :(
ASTERISK-02625: 487 Message not sent after receipt of CANCEL
ASTERISK-02626: Fax redirect does not always work
ASTERISK-02627: Monitor with option 'b' records silence when bridging Zap channels
ASTERISK-02628: CVS HEAD fails compile. Dundi problem?
ASTERISK-02629: [patch] wrong logging statement when sending voicemail alert email
ASTERISK-02630: help for http://bugs.digium.com/bug_report_page.php 403's
ASTERISK-02631: [patch] app_forkcdr output beautifications
ASTERISK-02632: [patch] ShowVars() function call OR try app_dumpchan.c
ASTERISK-02633: [patch] libpri compiles with wrong byte endian on FreeBSD
ASTERISK-02634: pbx_dundi.c doesn't compile out of the box on NetBSD
ASTERISK-02635: meetmeadmin/admin_exec() references null command pointer
ASTERISK-02636: 'asterisk -r -x "show channels"' often returns no channel information
ASTERISK-02637: [patch] SayUnixTime and SayNumber to say in British and Norwegian syntax
ASTERISK-02638: Fresh asterisk cvs won't compile - suspect lock.h
ASTERISK-02639: app_queue rings dynamic agents who are already on the phone
ASTERISK-02640: [PATCH] vmail.cgi patch to work with newer realtime config db name, and fixes taint problems when forwarding voicemails
ASTERISK-02641: [patch] ACK sent to wrong address
ASTERISK-02642: Country tones for Singapore
ASTERISK-02643: Country tones for Singapore
ASTERISK-02644: res_perl does not compile with CVS
ASTERISK-02645: Add preprocessor parsable ASTERISK_VERSION
ASTERISK-02646: [new_app] app_intercept.c
ASTERISK-02647: show channels crashes asterisk
ASTERISK-02648: [reqest] backgrounddetect also could listen for busy tones
ASTERISK-02649: Progress indicator must be optional in 'PROGRESS'
ASTERISK-02650: Crash in ast_queue_frame (channel.c)
ASTERISK-02651: dialog matching problems
ASTERISK-02652: asterisk "101 Dialog Establishement" error
ASTERISK-02653: rtp.c error - unknown RTP codec 127 received
ASTERISK-02654: Agents taking Calls
ASTERISK-02655: [patch] Add manager cmd - Action: Agents
ASTERISK-02656: port of pbx/pbx/dundi.c reset_global_eid() to FreeBSD
ASTERISK-02657: Asterisk Manager API (sockets)
ASTERISK-02658: ast_modem_pvt() uses memory before checking if it is valid
ASTERISK-02659: callfile, etc: Jump into extension without needing to have already established call
ASTERISK-02660: Add Israeli tone zone
ASTERISK-02661: request for dialing feature 'exten => (555)1234,1,app'
ASTERISK-02662: Sending SMS with no hangup supervision will hang the channel
ASTERISK-02663: False file-not-found error in voicemail
ASTERISK-02664: [patch] Chan Zap close bug
ASTERISK-02665: ZapRas/HDLCPPP is broken
ASTERISK-02666: RE bug # 0002688, New Country Indications for Singapore are likely wrong
ASTERISK-02667: [patch] Application UserEvent does not populate the 'body' parameter
ASTERISK-02668: [Patch] add a flag to app_dial to wait until someone has answer and confirm
ASTERISK-02669: Asterisk Makefile -DDEBUG_THREADS generating vast number of warnings.
ASTERISK-02670: Use the newly added FreeBSD MAC-address lookup for NetBSD
ASTERISK-02671: AudioCodes with last firmware couldnt register on Asterisk
ASTERISK-02672: joinempty does not work with members defined statically
ASTERISK-02673: [patch] Stack applications
ASTERISK-02674: Calls hung up when agent parks...
ASTERISK-02675: [patch] N+101 on failed playback
ASTERISK-02676: [patch] fix app_voicemail when using wav49
ASTERISK-02677: asterisk says telephone-event when apps want to hear PCMU.
ASTERISK-02678: Logging line in chan_local needs variables flipped
ASTERISK-02679: Asterisk sending PRI restarts every hour
ASTERISK-02680: latest v1-0 CVS chan_h323 doesn't want to compile on RH9
ASTERISK-02681: Parking a call crash the asterisk
ASTERISK-02682: [patch] Add distribution lists to app_voicemail
ASTERISK-02683: h323 calls gets recorded by voicemail application at double speed if voicemail format is .wav
ASTERISK-02684: comma "," gets changed to pipe "|"
ASTERISK-02685: [request] new cmd IF
ASTERISK-02686: Problem with multiple lines on Cisco phones.
ASTERISK-02687: Add pkg-config support to asterisk
ASTERISK-02688: No call pickup possible...
ASTERISK-02689: [patch] CLI command 'pri restart span <spannum>'
ASTERISK-02690: [patch]Allow AGI script to get HANGUPCAUSE variable
ASTERISK-02691: Asterisk hangs from time to time
ASTERISK-02692: cli summaries for h.323 show tokens and h.323 hangup swapped
ASTERISK-02693: [patch] Patch to make Asterisk work on Solaris
ASTERISK-02694: [patch] Chan SIP Transfer CRASH
ASTERISK-02695: zaptel/README is missing a prerequisite
ASTERISK-02696: [PATCH] DUNDi support for 1.0.2 release
ASTERISK-02697: [patch] zaptel/zaptel.conf.sample doesn't specify all zones
ASTERISK-02698: [patch] logger.c casts pthread_self() to long
ASTERISK-02699: nonce generation errors
ASTERISK-02700: SIP hold/transfer fails
ASTERISK-02701: [patch] Give warning on compile with old libpri and new QSIG
ASTERISK-02702: [patch] Say number for portuguese is wrong
ASTERISK-02703: [request] when dial connects, fork/reset the CDR.
ASTERISK-02704: Zaptel documentation doesn't say anything on how to set the card to an E1
ASTERISK-02705: No queue announcements when MOH is replaced by ringing
ASTERISK-02706: Festival not working for Asterisk 1.0
ASTERISK-02707: Wrong Request URI - RFC3261
ASTERISK-02708: [patch] make table name for "cdr" table configurable in cdr_odbc and cdr_odbc module unloadable
ASTERISK-02709: Incoming calls handling, FastStart & H245 state
ASTERISK-02710: [patch] Add depend support to h323/ast_h323
ASTERISK-02711: [patch] libpri debugging cleanup
ASTERISK-02712: [patch] PRI progress indicator support
ASTERISK-02713: [patch] PRI redirecting number IE support
ASTERISK-02714: app_record is broken saying that filename argument is not present
ASTERISK-02715: [patch] Little code cleanups for current CVS-HEAD
ASTERISK-02716: [patch] Passing REDIRECTING NUMBER IE on PRI outgoing calls
ASTERISK-02717: [patch] Don't show partially-filled strings over remote console (asterisk -r)
ASTERISK-02718: [patch] More output for debugging purposes
ASTERISK-02719: Patch to app_disa for stutter-dialtone and response / digittimeout.
ASTERISK-02720: presence/hint as an application
ASTERISK-02721: Erroneous warning Dial argument takes format (technology1/[device:]number1&technology2/[device:]number2...|optional timeout)
ASTERISK-02722: after asterisk upgrade it reports error and doesn't load anymore.
ASTERISK-02723: getting a (matching?) query while dundi debug is on crashes asterisk
ASTERISK-02724: Incomming caller*id presentation
ASTERISK-02725: Asterisk Crash/Core Dump
ASTERISK-02726: IAX2 trunked channel cannot recover from loss of packet containing 'first full voice frame'?
ASTERISK-02727: h323 calls gets recorded by voicemail application at double speed if voicemail format is .wav
ASTERISK-02728: [patch] VMAuthenticate
ASTERISK-02729: [patch] STREAM FILE supports a timeout in res_agi.c
ASTERISK-02730: Make channel-vars readable with pbx_builtin_getvar_helper
ASTERISK-02731: Trouble with music on hold when fname_base contains special characters
ASTERISK-02732: Unknown RTP codec 72 received
ASTERISK-02733: [patch] German syntax for say.c and it's neighborhood; Creation of say_enumeration (1st,2nd,...,101st,...,last)
ASTERISK-02734: Syntax error before * token in file included from chan_phone.c:37:
ASTERISK-02735: Zaptel crashes when unloaded, then reloaded and alarm cleared
ASTERISK-02736: [PATCH] allow zap gain and echo params to be twiddled on the fly from console
ASTERISK-02737: [request] automatic disabling echo cancellation when data connection is on a call
ASTERISK-02738: chan_h323 will not compile
ASTERISK-02739: [patch] Generic Digits preliminary support
ASTERISK-02740: [patch] - Fix correct output of sendpage debug
ASTERISK-02741: compilation error in chan_h323 in 1.0.2 stable version
ASTERISK-02742: variables are lost in a blind transfer
ASTERISK-02743: libpri no longer detects remote answer
ASTERISK-02744: SIP Notify / Message waiting does not conform to RFC.
ASTERISK-02745: AMIS networking
ASTERISK-02746: Callier ID Name is lost when calling from SIP channel to Zap channel.
ASTERISK-02747: [PATCH] added support for pins to astcc.
ASTERISK-02748: [PATCH] New agi script ASTPP
ASTERISK-02749: Teach zaptel's Makefile to _actually_ install into arbitary locations
ASTERISK-02750: [patch] Do not use call progress analysis on PRI links
ASTERISK-02751: [patch] Frame debugging enhancements
ASTERISK-02752: oneway audio with iaxy due to CODEC mismatch caused by iax.conf
ASTERISK-02753: SIP REGISTER ignores 'authuser' setting
ASTERISK-02754: Asterisk build fails on Darwin/OSX
ASTERISK-02755: temporarily set SIP UA string on CLI
ASTERISK-02756: WCUSB driver doesnt work
ASTERISK-02757: Cisco CID blocking cause crash
ASTERISK-02758: Change HOSTCC to CC
ASTERISK-02759: [patch] app behavioral modifications
ASTERISK-02760: [patch] fix seeding verbose
ASTERISK-02761: Spelling error
ASTERISK-02762: Request for decimal rates
ASTERISK-02763: Request for decimal rates
ASTERISK-02764: Supervised transfer will result into called party hearing moh
ASTERISK-02765: [patch] Add timezone support for IAX date/time Information Element
ASTERISK-02766: config from db via include
ASTERISK-02767: chan_sip segfaults on incoming call from FWD.
ASTERISK-02768: Include in extensions.conf seems to be broken with asterisk-1.0.1 and 1.0.2
ASTERISK-02769: Voicemail doesn't work with mailbox names beginning with "u"
ASTERISK-02770: Voicemail doesn't work with mailbox names beginning with "u"
ASTERISK-02771: [patch] correctly reinit a variable and other more substantial modifications to MARK2 echo canceller
ASTERISK-02772: Incomplete CDR record when originating call from manager
ASTERISK-02773: [patch] Support for Note2 Table 4-3/Q.931 (some information elements may be repeated)
ASTERISK-02774: [patch] clean up a bit in chan_sip
ASTERISK-02775: [PATCH] app_vmoutcall
ASTERISK-02776: [patch] Allow globbing in #include on config files
ASTERISK-02777: [patch] Support for disabling detection of certain call progress tones
ASTERISK-02778: goertzel dsp was off-by-one
ASTERISK-02779: outgoing spool directory dials number correctly then need way to dial extension like D(123) from app_dial
ASTERISK-02780: add option to app_chanisavail for "in use"
ASTERISK-02781: [patch] expose dsp->tstate and dsp->count if needed.
ASTERISK-02782: [PATCH] chan_phone.c does not compile with 2.6 kernel headers
ASTERISK-02783: [PATCH] FD not close after getting default local EID
ASTERISK-02784: Asterisk 1.02 quits upon launch on Darwin/OSX
ASTERISK-02785: pedantic=yes makes calls to voip providers impossible
ASTERISK-02786: [patch] Patterns may include -, but extensions may not
ASTERISK-02787: [patch] Fullcontact fixes
ASTERISK-02788: [patch] Make mailbox check time configurable
ASTERISK-02789: [patch] New app in chan_sip: sipgetheader()
ASTERISK-02790: expose timing functions from pbx.c for use in other appliactions
ASTERISK-02791: [patch] Italian date syntax for say.c
ASTERISK-02792: Blind call transfers on SIP channels don't work from within AGI app
ASTERISK-02793: [patch] Italian date syntax for app_voicemail.c
ASTERISK-02794: Faster ADPCM and G726-32 codec code
ASTERISK-02795: User control of specific authentication methods
ASTERISK-02796: If voicemail.conf #includes additional files, when you try to change PIN it doesn't work.
ASTERISK-02797: [patch] SipAddHeader() Application
ASTERISK-02798: [patch] - Code to allow reverse polarity to indicate a hangup on the channel
ASTERISK-02799: language=de can't chage the language in sip.conf
ASTERISK-02800: [patch] allow setting permissions so vmail.cgi can read vmailbox
ASTERISK-02801: SIP messages contain 0.0.0.0 as IP for Asterisk
ASTERISK-02802: [patch] Preset Channel Vars In Users/Friends In iax.conf/sip.conf
ASTERISK-02803: [patch] it allows to send the response 180 Ringing even if has been already sended '183 Progress Response'
ASTERISK-02804: Type of Number not in CVS available as claimed
ASTERISK-02805: [patch] fix building on older linux systems
ASTERISK-02806: [patch] Formatting and additional comments
ASTERISK-02807: [request] Keeping Asterisk in the Channel Path using DUNDi and notransfer=yes
ASTERISK-02808: [patch] file.c const const warning
ASTERISK-02809: [patch] make res_crypto less chatty under valgrind.
ASTERISK-02810: [patch] Line 227 & 228 in res_musiconhold.c wtf are we closing really?
ASTERISK-02811: [patch] Outbound proxy support
ASTERISK-02812: bad dependency on order of contexts in sip.conf
ASTERISK-02813: [patch] cli.c tweaks
ASTERISK-02814: Crash when terminating several calls at the same time
ASTERISK-02815: [branch] RTCP-support
ASTERISK-02816: [patch] add -e option to exec something after bootup
ASTERISK-02817: [patch] deadlock H.323
ASTERISK-02818: [patch] libiax2 updates: CNG, MSVC, XFER, ALIGN, codecs, misc
ASTERISK-02819: [patch] leavewhenempty doens't work
ASTERISK-02820: [patch] GET OPTION in AGI
ASTERISK-02821: [patch] See any playing files when STREAM FILE
ASTERISK-02822: have sip not consider port when matching a peer
ASTERISK-02823: [patch] Added brazilian tones to zonedata.c
ASTERISK-02824: [patch] Call progress detection for Brazil
ASTERISK-02825: SIP channel rings after answer on some calls from Polycom
ASTERISK-02826: [request + patch?] SIP REGISTER timeout setting wanted
ASTERISK-02827: sip invite retry with wrong password
ASTERISK-02828: asterisk UAC function need to send 487 after get CANCEL in sip
ASTERISK-02829: Dial fails
ASTERISK-02830: [patch] app_dial use of dial options disallows unlimited timeout without warning.
ASTERISK-02831: [patch] Support MGCP distinctive ring
ASTERISK-02832: Retry poll() on EINTR in ast_waitfor_nandfds()
ASTERISK-02833: [patch] fix silence detection in app_record
ASTERISK-02834: CVS 1-0 and 1-0-2 mismatch (libpri features/function)
ASTERISK-02835: Connecting queued person to agent fails
ASTERISK-02836: [request] Proper pattern matching when using chan_phone
ASTERISK-02837: DTMF Stops working in Voicemail
ASTERISK-02838: [patch] chan_mgcp is not functional in CVS HEAD
ASTERISK-02839: [patch] fix UK CallerID on outward FXS port
ASTERISK-02840: [patch] [v1-0] Backport locking changes from CVS HEAD + formating/spacing
ASTERISK-02841: segfault if head caller times out of queue with moh running
ASTERISK-02842: [PATCH] SIP header continuation line parsing not conformant to RFC
ASTERISK-02843: [patch] Mailbox getting lost with realtime config
ASTERISK-02844: [patch] Add a continue option to WaitExten
ASTERISK-02845: 'i' extension doesnt match invalid dialled extensions, only menu responses
ASTERISK-02846: Dropping voice to exceptionally long queue
ASTERISK-02847: if using complex codec and inband dtmf, ast_dsp_process reports wrong codec in error message
ASTERISK-02848: Use counter not decremented after timeout in INVITE
ASTERISK-02849: [patch] Misc fixes
ASTERISK-02850: dsp.c hangup detection on x100p
ASTERISK-02851: Request for informed comment on behaviour of wctdm driver
ASTERISK-02852: [patch] Asterisk sends repeated rtp seq number with rfc2833 dtmf
ASTERISK-02853: I get one way audio when calling from iaxy(ULAW) to zaptel(ALAW)
ASTERISK-02854: [patch] RTP keepalive for SIP
ASTERISK-02855: [patch] Improve app_dial M(macro) stuff
ASTERISK-02856: [patch] leading + and - in Goto/GotoIf to go with priority n stuff
ASTERISK-02857: app_while
ASTERISK-02858: [patch] "make samples" always overwrites current configuration
ASTERISK-02859: V23 CID hangs on polarity reversal that is not a RING
ASTERISK-02860: Incorrect parsing of remote-party-ID
ASTERISK-02861: [patch] Multi-Line Comments In Config Files
ASTERISK-02862: [patch] MacroIf()
ASTERISK-02863: [patch] Override native transfers by a dialplan predefined extension
ASTERISK-02864: Call in queue tries twice with 'n' option
ASTERISK-02865: [patch] Reset variables at SIP reload
ASTERISK-02866: [patch] Re-use registration credentials
ASTERISK-02867: SIP to H323 Bridge Issue
ASTERISK-02868: [patch] iax peers are always reported as lagged on asterisk manager
ASTERISK-02869: FXO interface stops handling outgoing calls following polarity reversal when using V23 signalling
ASTERISK-02870: Remove the new line in app_alarmreceiver.c
ASTERISK-02871: [patch] Supports filaname with a dot in app_record.c
ASTERISK-02872: [patch] Incoming SIP "Distinctive Ring" detection using Alert-Info header
ASTERISK-02873: NAT on REGISTER
ASTERISK-02874: [patch] var_val in res_config_odbc.c
ASTERISK-02875: [patch] allow zapscan to scan based on channel GROUP
ASTERISK-02876: [patch] do not increment ts value on ast_rtp_senddigit
ASTERISK-02877: [patch] Reload dynamic queue members on restart
ASTERISK-02878: [new_app] WaitIVR
ASTERISK-02879: contact header patch
ASTERISK-02880: endless loop due to ast_search_dns() taking too long
ASTERISK-02881: [request] Implement T.38 as a codec
ASTERISK-02882: [patch] Let Background() take mutiple files
ASTERISK-02883: Asterisk does not register with SIP proxies unless a "sip reload" command is issued
ASTERISK-02884: [patch] fix MEETMESECS in app_meetme
ASTERISK-02885: chan_mgcp does not send correct ip in sdp until reloaded
ASTERISK-02886: [patch] SIP channel goes ZOMBIE when transfering call on IAX to MeetMe
ASTERISK-02887: Non-existant channel type to chan_features crashes Asterisk.
ASTERISK-02888: [patch] fix memory leak in cdr_odbc.c
ASTERISK-02889: Previous locking improvements in chan_mgcp need to be revised (???)
ASTERISK-02890: [patch] Call forwarding to self with chan_sip causes loop.
ASTERISK-02891: IAX2 transfers fails when one party is behind nat and iax port isn't dnat'ed inside
ASTERISK-02892: [patch] missing ast_destroy(cfg)
ASTERISK-02893: [patch] when disallow=all and allow=all in [general], all other codec settings in peers do not workand no sound heard.
ASTERISK-02894: app_runexten - Run a give extension from CLI
ASTERISK-02895: [patch] Run script upon registration chan_sip.c
ASTERISK-02896: [patch] compact sip headers
ASTERISK-02897: Repeated Zaptel Logger Entry
ASTERISK-02898: Did the new cvs update break oh323???
ASTERISK-02899: [patch] make SQL table name for "cdr" table configurable in cdr_pgsql
ASTERISK-02900: [patch] Meetme Conference Cloaking and Status Display
ASTERISK-02901: EAGI fails somehow when a long distance caller connects
ASTERISK-02902: make progdocs requires graphviz
ASTERISK-02903: [patch] press a key, record a call. Configured via dialplan variables.
ASTERISK-02904: E1 PRI won't bring b-channels up
ASTERISK-02905: canceled voicemail emails corrupt wav
ASTERISK-02906: [patch] Specify the machine in the safe_asterisk
ASTERISK-02907: CVS Head doesnt compile!
ASTERISK-02908: Suspend in tcsh, kill %1 *twice*, then fg gives core dump
ASTERISK-02909: [patch] print the number of applications registered
ASTERISK-02910: [PATCH] cdr_dumper_php
ASTERISK-02911: festival source-code patching not needed anymore
ASTERISK-02912: SIP response code handling
ASTERISK-02913: [patch] app_record with silence detection gets read format stuck
ASTERISK-02914: [patch] Zaptel does not detect hangup in Singapore
ASTERISK-02915: [patch] Add counters in the show dialplan
ASTERISK-02916: pbx_realtime.c - Docs and Minor Tweaks
ASTERISK-02917: Early hangup during MGCP transfer (like unattended transfer) crashes running asterisk
ASTERISK-02918: [patch] Control codec priorities in IAX2
ASTERISK-02919: no tabs for show channels concise
ASTERISK-02920: [patch] new version of app_sms.c
ASTERISK-02921: sip.conf parsing error
ASTERISK-02922: chan_zap.c always tries to use channel 24 as D-channel
ASTERISK-02923: Recent multiline comment if conf files conflicts with sample sip.conf
ASTERISK-02924: [patch] H323 channel: in callback send_digit() argument call_token missing
ASTERISK-02925: Pbx_realtime - SQL Fetch error!
ASTERISK-02926: MWI system is currently polling based, this makes it event based and fixed ODBC storage issues
ASTERISK-02927: ALERT_INFO and VXML_URL not sent
ASTERISK-02928: [PATCH] Send DLCX to MGCP gateway when channel is already dead in Asterisk
ASTERISK-02929: Strange errors on incoming call
ASTERISK-02930: [patch] Realtime extensions not resolving variables
ASTERISK-02931: app_dial leaks frames in some error conditions
ASTERISK-02932: [patch] CLI 'database showkey <keytree>'
ASTERISK-02933: [patch] adds shortcutting when selecting translation paths
ASTERISK-02934: will not load due to strdupa call added to chan_sip.c
ASTERISK-02935: app_dial DIALSTATUS is always CHANUNAVAIL when not answered
ASTERISK-02936: Memory leak in ast_expr.y
ASTERISK-02937: Reload from CLI while using MeetMe causes crash
ASTERISK-02938: [patch] Hungarian tones (zaptel+indications)
ASTERISK-02939: using outgoing spool to CONSOLE/dsp and agi sends hangup to other channel
ASTERISK-02940: faxdetect fails on Asterisk CVS-HEAD-12/07/04-21:50:06
ASTERISK-02941: [patch] Patch to allow the directory to exit when user presses "0".
ASTERISK-02942: app_queue and chan_local maximum usage?
ASTERISK-02943: [patch] noload'able res_musiconhold.so
ASTERISK-02944: fmtp payload header
ASTERISK-02945: chan_agent.so and chan_local.so use count
ASTERISK-02946: Crash on chan_h323
ASTERISK-02947: MusicOnHold and Transferring a call does not work in a queue
ASTERISK-02948: Asterisk crashes as soon as there are more than 10 calls on the TE110P
ASTERISK-02949: [patch] MWI subscribe error when type=friend
ASTERISK-02950: [request] Console/dsp extension does not work with AC97 (48kHz stereo-only) audio chips
ASTERISK-02951: New batch of phrases for Allison
ASTERISK-02952: IAX2 to IAX2 huge delays
ASTERISK-02953: [patch] Fix memory leak in ast_expr.y
ASTERISK-02954: [patch] res_config_mysql.c causes core dump
ASTERISK-02955: Using SetVar Incorrectly Causes Asterisk Process to Crash
ASTERISK-02956: Channel in strange state after transferring 2nd call
ASTERISK-02957: Can meetme and voice mail integrate
ASTERISK-02958: [patch] app_read Addition
ASTERISK-02959: bad clicking/popping only with SMP kernel 2.6.x
ASTERISK-02960: Report IAX2 Frame_Text to send Hangup Cause from Zap channel
ASTERISK-02961: [patch] Polish tones (zaptel & asterisk indications)
ASTERISK-02962: adding SIPURI to predefined channel variables
ASTERISK-02963: no audio
ASTERISK-02964: Umm..whats with all the ODBC stuff inside app_voicemail?
ASTERISK-02965: [PATCH] allow contexts to be repeated in multiple files, so they add together
ASTERISK-02966: GR-303 Compatibility issues with AFC AccessMAX / UMC1000
ASTERISK-02967: allow "category" to be assigned to messages
ASTERISK-02968: Changes to language handling/minor bug
ASTERISK-02969: [PATCH] remove need for voicemail-related symlinks in sounds directory
ASTERISK-02970: utils/smsq.c compile errors on FreeBSD 4.10
ASTERISK-02971: Timeout problem with CallerID via DTMF/Polarity on TDM400P
ASTERISK-02972: [patch] move process_quotes_and_slashes into pbx.c rename to ast_process_quotes_and_slashes
ASTERISK-02973: [patch] cdr_pgsql.c this has been bugging me.
ASTERISK-02974: [patch] ODBC Realtime extensions switch requires MySQL
ASTERISK-02975: [patch] check the RTP version for find invalid frame
ASTERISK-02976: [patch] Channels get stuck in the parking lot
ASTERISK-02977: [patch] Send Q.931 cause codes
ASTERISK-02978: autofallthrough is incompatible with IVR menus; should default to "no"
ASTERISK-02979: [patch] Move ast_app_has_voicemail and ast_app_messagecount to app_voicemail
ASTERISK-02980: [patch] Stop MOH on hangup rather than after the channel is destroyed
ASTERISK-02981: [patch] Clean up a couple of things left behind by "make clean"
ASTERISK-02982: wrong subroutine name in chan_zap.c
ASTERISK-02983: [patch] queue priority (weight)
ASTERISK-02984: res_config_mysql.c Crashes BADLY
ASTERISK-02985: Crash on masquerading <ZOMBIE> channels
ASTERISK-02986: Cisco's RTP codec type 100
ASTERISK-02987: Read() exists non-zero on timeout
ASTERISK-02988: Agent Hangup Crashes Asterisk
ASTERISK-02989: macros cannot exit on */# keys like regular contexts
ASTERISK-02990: [PATCH] add MacroExit application
ASTERISK-02991: [patch] Make ast_test_flag and friends a macro
ASTERISK-02992: [patch] Wrong flags passed to glob, and fix for solaris
ASTERISK-02993: [patch] Correction to markster's tweak for quad_t
ASTERISK-02994: [patch] utils/Makefile doesn't work on Solaris
ASTERISK-02995: mkdep script missing
ASTERISK-02996: glob call in config.c causes seg fault
ASTERISK-02997: cdr_csv lets write when cdr_odbc loses connection with a remote database....
ASTERISK-02998: RealTime + regexten does not work
ASTERISK-02999: Fails to compile on Linux FC3 (latest CVS)